webrtc_m130/modules/audio_device/audio_device_impl.h
Artem Titov eeae962997 Reland "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
This reverts commit 69c8d3c843326aff9dee32cc639741c1cd7f8ae9.

Reason for revert: Reland with a fix

Original change's description:
> Revert "Migrate TestAudioDeviceModule on AudioDeviceModuleImpl"
>
> This reverts commit e42bf81486d2f08b6dcbf1442287202e937ce52b.
>
> Reason for revert: Breaks iOS simulator bots and thus blocks chromium roll, https://chromium-review.googlesource.com/c/chromium/src/+/4433814
>
> Original change's description:
> > Migrate TestAudioDeviceModule on AudioDeviceModuleImpl
> >
> > Bug: b/272350185
> > Change-Id: Ia3d85d6fa3b0d4809e987a39d60d3eb022687132
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300363
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39877}
>
> Bug: b/272350185
> Change-Id: I1e3b542fc1278797f283afedeae01cbb7412d353
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301701
> Commit-Queue: Jeremy Leconte <jleconte@google.com>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Auto-Submit: Christoffer Jansson <jansson@google.com>
> Owners-Override: Christoffer Jansson <jansson@google.com>
> Cr-Commit-Position: refs/heads/main@{#39881}

Bug: b/272350185
Change-Id: I809466306b2e1fd54c44b90311059c98a53ef8ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301704
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39936}
2023-04-24 14:42:08 +00:00

192 lines
6.9 KiB
C++

/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_
#if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
#include <stdint.h>
#include <memory>
#include "api/task_queue/task_queue_factory.h"
#include "modules/audio_device/audio_device_buffer.h"
#include "modules/audio_device/include/audio_device.h"
namespace webrtc {
class AudioDeviceGeneric;
class AudioManager;
class AudioDeviceModuleImpl : public AudioDeviceModuleForTest {
public:
enum PlatformType {
kPlatformNotSupported = 0,
kPlatformWin32 = 1,
kPlatformWinCe = 2,
kPlatformLinux = 3,
kPlatformMac = 4,
kPlatformAndroid = 5,
kPlatformIOS = 6,
// Fuchsia isn't fully supported, as there is no implementation for
// AudioDeviceGeneric which will be created for Fuchsia, so
// `CreatePlatformSpecificObjects()` call will fail unless usable
// implementation will be provided by the user.
kPlatformFuchsia = 7,
};
int32_t CheckPlatform();
int32_t CreatePlatformSpecificObjects();
int32_t AttachAudioBuffer();
AudioDeviceModuleImpl(AudioLayer audio_layer,
TaskQueueFactory* task_queue_factory);
// If `create_detached` is true, created ADM can be used on another thread
// compared to the one on which it was created. It's useful for testing.
AudioDeviceModuleImpl(AudioLayer audio_layer,
std::unique_ptr<AudioDeviceGeneric> audio_device,
TaskQueueFactory* task_queue_factory,
bool create_detached);
~AudioDeviceModuleImpl() override;
// Retrieve the currently utilized audio layer
int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
// Full-duplex transportation of PCM audio
int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
// Main initializaton and termination
int32_t Init() override;
int32_t Terminate() override;
bool Initialized() const override;
// Device enumeration
int16_t PlayoutDevices() override;
int16_t RecordingDevices() override;
int32_t PlayoutDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
int32_t RecordingDeviceName(uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]) override;
// Device selection
int32_t SetPlayoutDevice(uint16_t index) override;
int32_t SetPlayoutDevice(WindowsDeviceType device) override;
int32_t SetRecordingDevice(uint16_t index) override;
int32_t SetRecordingDevice(WindowsDeviceType device) override;
// Audio transport initialization
int32_t PlayoutIsAvailable(bool* available) override;
int32_t InitPlayout() override;
bool PlayoutIsInitialized() const override;
int32_t RecordingIsAvailable(bool* available) override;
int32_t InitRecording() override;
bool RecordingIsInitialized() const override;
// Audio transport control
int32_t StartPlayout() override;
int32_t StopPlayout() override;
bool Playing() const override;
int32_t StartRecording() override;
int32_t StopRecording() override;
bool Recording() const override;
// Audio mixer initialization
int32_t InitSpeaker() override;
bool SpeakerIsInitialized() const override;
int32_t InitMicrophone() override;
bool MicrophoneIsInitialized() const override;
// Speaker volume controls
int32_t SpeakerVolumeIsAvailable(bool* available) override;
int32_t SetSpeakerVolume(uint32_t volume) override;
int32_t SpeakerVolume(uint32_t* volume) const override;
int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
// Microphone volume controls
int32_t MicrophoneVolumeIsAvailable(bool* available) override;
int32_t SetMicrophoneVolume(uint32_t volume) override;
int32_t MicrophoneVolume(uint32_t* volume) const override;
int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
// Speaker mute control
int32_t SpeakerMuteIsAvailable(bool* available) override;
int32_t SetSpeakerMute(bool enable) override;
int32_t SpeakerMute(bool* enabled) const override;
// Microphone mute control
int32_t MicrophoneMuteIsAvailable(bool* available) override;
int32_t SetMicrophoneMute(bool enable) override;
int32_t MicrophoneMute(bool* enabled) const override;
// Stereo support
int32_t StereoPlayoutIsAvailable(bool* available) const override;
int32_t SetStereoPlayout(bool enable) override;
int32_t StereoPlayout(bool* enabled) const override;
int32_t StereoRecordingIsAvailable(bool* available) const override;
int32_t SetStereoRecording(bool enable) override;
int32_t StereoRecording(bool* enabled) const override;
// Delay information and control
int32_t PlayoutDelay(uint16_t* delayMS) const override;
bool BuiltInAECIsAvailable() const override;
int32_t EnableBuiltInAEC(bool enable) override;
bool BuiltInAGCIsAvailable() const override;
int32_t EnableBuiltInAGC(bool enable) override;
bool BuiltInNSIsAvailable() const override;
int32_t EnableBuiltInNS(bool enable) override;
// Play underrun count.
int32_t GetPlayoutUnderrunCount() const override;
#if defined(WEBRTC_IOS)
int GetPlayoutAudioParameters(AudioParameters* params) const override;
int GetRecordAudioParameters(AudioParameters* params) const override;
#endif // WEBRTC_IOS
#if defined(WEBRTC_ANDROID)
// Only use this acccessor for test purposes on Android.
AudioManager* GetAndroidAudioManagerForTest() {
return audio_manager_android_.get();
}
#endif
AudioDeviceBuffer* GetAudioDeviceBuffer() { return &audio_device_buffer_; }
int RestartPlayoutInternally() override { return -1; }
int RestartRecordingInternally() override { return -1; }
int SetPlayoutSampleRate(uint32_t sample_rate) override { return -1; }
int SetRecordingSampleRate(uint32_t sample_rate) override { return -1; }
private:
PlatformType Platform() const;
AudioLayer PlatformAudioLayer() const;
AudioLayer audio_layer_;
PlatformType platform_type_ = kPlatformNotSupported;
bool initialized_ = false;
#if defined(WEBRTC_ANDROID)
// Should be declared first to ensure that it outlives other resources.
std::unique_ptr<AudioManager> audio_manager_android_;
#endif
AudioDeviceBuffer audio_device_buffer_;
std::unique_ptr<AudioDeviceGeneric> audio_device_;
};
} // namespace webrtc
#endif // defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
#endif // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H_